A conference call is a call between three or more callers/called parties, where each party can hear each of the other parties (the number of conferencing parties is often limited, and in some systems, the number of simultaneous talkers may also be limited). Conferencing capabilities exist in the PSTN (Public Switched Telephone Network), where remote caller's voices are mixed, e.g., at the central office, and then sent to a conference participant over their single line. Similar capabilities can be found as well in many PBX (Private Branch Exchange) systems.
Packet-switched networks can also carry real time voice data, and therefore, with proper configuration, conference calls. Voice over IP (VoIP) is the common term used to refer to voice calls that, over at least part of a connection between two endpoints, use a packet-switched network for transport of voice data. VoIP can be used as merely an intermediate transport media for a conventional phone, where the phone is connected through the PSTN or a PBX to a packet voice gateway. But other types of phones can communicate directly with a packet network. IP (Internet Protocol) phones are phones that may look and act like conventional phones, but connect directly to a packet network. Soft phones are similar to IP phones in function, but are software-implemented phones, e.g., on a desktop computer.
Since VoIP does not use a dedicated circuit for each caller, and therefore does not require mixing at a common circuit switch point, conferencing implementations are somewhat different than with circuit-switched conferencing. In one implementation, each participant broadcasts their voice packet stream to each other participant—at the receiving end, the VoIP client must be able to add the separate broadcast streams together to create a single audio output. In another implementation, each participant addresses their voice packets to a central MCU (Multipoint Conferencing Unit). The MCU combines the streams and sends a single combined stream to each conference participant.